Hi !

I'm not sure if it's the right list but I hope someone can help me !

I am setting up a SIP VoIP gateway using Asterisk and a Digium E100P
card plugged on a Swisscom PRI ISDN.

I setup my card using ccs framing, hdb3 coding with crc4 checking and
euroisdn switch type.

Everything works well with incoming calls, my Grandstream SIP phones
ring and answer calls, show right CID and so on, but when dialling out
from SIP to PSTN, it looks like it works, asterisk shows the correct
things, says number is ringing, but nothing happens on the remote PSTN
phone.

Asterisk has no "tone zone" for Switzerland, means no DTMF info. I tried
several other countries DTMF settings with no success thinking it might
be the cause.

Any suggestion ? Do you know Swisscom's DTMF settings ? Similar to other
countries ??

Thanks.



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